Abstract: The invention relates to a method for transmitting data having a reduced bandwidth wherein the amount of data intended for transfer over a data transfer network is reduced by a reduction method, wherein the type of reduction performed is signaled by a source switching device to a terminal switching device in the data transfer network, such that in the terminal switching device, transferred data is processed using a partner method to the reduction method in order to recover the original data.

Abstract: The N peaks are selected from a delay profile in order starting from the highest peak involved in the delay profile, the path level and path timing are determined for each of the selected peaks, and the identical paths individually responsible for different delay profiles are detected by comparing the path timings calculated for the preceding time and the path timings calculated for the present time. Based of the path levels calculated for the preceding time and the path levels calculated for the present time, averaged path levels are computed on a path-by-path basis for the present time, a predetermined threshold is applied to the computed average path levels, and the paths to be allocated to the fingers are selected. Of the path timings that have been obtained for respective selected paths, the latest path timings are allocated to the fingers as synchronization timings.

Abstract: A node of a telecommunications network comprises a first function configured to perform a first operation on a first portion of a packet handled by the node and a second function configured to perform a second operation on a second portion of the packet. The first function and the second function are configured to employ a shared transaction for operating on the packet whereby, by virtue of the shared transaction, after performance of the first operation and the second operation, the packet has less overhead attributable to the first function and the second function than if the shared transaction had not been employed in performance of the first operation and the second operation.

Abstract: A method and system for the synchronization of the transmission of digitally converted analog modem signals between two or more modems using either RF over-the-air or A/C power line communication channels. This invention provides for modem speed optimization by minimizing or eliminating clock variance between the modems in a digital communication system. A fixed delay is created and maintained during modem operation. This invention is also adapted to function in a full-duplex communication system maintaining a fixed delay in both directions.

Abstract: Due to bandwidth constraints on the wireless link in an IP network, it is useful to compress the headers so as to maximize the utilization of the link. There exists Header Compression algorithms that make use of the similarity in consecutive headers in a packet flow to compress these headers. In this document, a novel header compression scheme was introduced that makes use of the similarity in consecutive flows from or to a given mobile terminal to compress these headers. Using information theory, the optimal gain to be expected from the use of such a scheme was analyzed. A model was defined for the distribution of the connections of a single user over the address space. The compression scheme was evaluated with respect to this model and to actual internet data traces. The scheme is complementary and the benefits are additional to the traditional approach to header compression. However, the scheme outperforms current schemes with respect to actual internet traces.

Abstract: A network apparatus for performing functions to enhance performance of a communication network is provided. The network apparatus includes a spoofing module that is configured to selectively spoof a multiple connections associated with a multiple hosts based upon corresponding spoofing criteria and to provide local acknowledgement of received messages over the connections. Additionally, the network apparatus includes a connection module that multiplexes the connections over a common backbone connection, and a prioritization module that prioritizes access to the backbone connection based upon prioritization criteria. Further, the network apparatus includes a path selection module that determines a path to transmit the received messages based upon path selection criteria. The spoofing module allocates a connection control block corresponding to a spoofed connection. Each of the connection control blocks stores information related to the connections.

Abstract: The invention comprises a system and method for correcting jitter and frame erasure in packet voice communication systems with out severely affecting the voice quality of the signal to be easily noticed by a listener. A packet is retrieved from a buffer, a pitch of the packet is determined, and the pitch of the packet is processed in a manner determinative of whether a next packet has arrived.

Abstract: In a wireless packet switching telecommunications network, voice services are provided by having a compressor/decompressor in each mobile station to provide each voice packet with a compressed header. Voice data and signalling data are sent separately and in different data formats to the air interface. The compressed header may be an M bit and a cyclically reset timeclick_number, which is decompressed by use of a wallclock which counts reset cycles to reinstate the voice packet headers. Alternatively, RTP agents are provided at the compression and decompression points, and voice packets are sent without headers over a “high quality” network such as a frame relay or ATM network. Compression state of a voice packet header can be established by sending call setup information over an out-of-band channel between compression points in a mobile station and in the network.

Abstract: A method and system for resetting nodes in a communication system that include at least a first node and a second node. The establishment for communication settings is initiated between the first node and the second node. At least part of communications settings is stored in the first node. An indication is sent from the second node to the first node to reset the stored communication settings in the first node. The communication settings are reset in the first node.

Abstract: A method, apparatus, system, and computer software for compressing and decompressing a message for transmission. The method of compressing a text message for transmission may include parsing text strings and encoding numerical values with a binary representation and analyzing values of the text strings and populating a session specific codebook with partial strings from the values.

Abstract: A device and method for communication of speech packets over a packet-switched network allows for a greater channel reallocation delay. Initial speech packets may be buffered during a channel reallocation delay and sent through an access medium when a channel is granted. A media access controller may transmit the buffered speech packets through the access medium at a rate exceeding a speech encoding rate. At the receiving user equipment, the initial speech packets are decoded and buffered. Speech signals representative of the initial speech packets may have a shortened time period to compensate at least in part for the channel reallocation delay. Decoded speech packets may be processed using a rate matching process at a rate which initially exceeds the speech encoding rate which may be gradually decreased to approximately the speech encoding rate. The rate matching may include dynamic time warping to substantially preserve attributes of the original speech.

Abstract: In a data compression system for compressing an original time series data with a various waveform, a compression unit compresses the original time series data without damaging characteristics of waveform information in the various waveform. The waveform information includes a signal with a various change, especially, a sudden change of signal. An encoding unit encodes the compressed time series data so as to generate a compression code.

Abstract: The present invention is a system and method for enabling multicast synchronization of initially unicasted content. Multiple unicast streams are synchronized in order to convert the unicast streams into a multicast stream. Each unicast stream may be accelerated or slowed down in relation to a reference stream to a common point within each stream upon which the unicast streams are replaced by a multicast stream of the same content.

Abstract: A method and apparatus for improving the transmission quality in a packet-oriented data transmission network, wherein time-critical data is transmitted over a packet-oriented data transmission network having a number of communications equipment units which are networked to one another.

Abstract: Under the situation in a network including a transmitter and a receiver in which a transmitter converts a non-compressed packet to be transmitted as a full-header packet including a full header or a header-compressed packet including a compressed header, and sends the converted packet to the receiver, the transmitter sends at least an important packet in the non-compressed packet to be transmitted as a full-header packet. The important packet means for example, a packet containing the data serving an important role when the data terminal eventually receiving each packet reproduces the audio and/or image in accordance with the data in each packet.

Abstract: A digital microelectronic circuit comprises a clocked data-processing unit (1) and a converting unit (2) which reads in data present at the output of the data-processing unit, performs a predetermined converting operation on the data and passes on the converted data. The converting unit is realized in an asynchronous logic circuit, such that the period of time for performing the converting operation is shorter than the shortest time interval to the next change of the data present at the output of the data-processing unit. In this way, fast, serial synchronous processes can be parallelized from the point of view of the slow synchronous system in synchronous systems which are slow relative thereto by using asynchronous logics, without a further high-frequency clock system being required.

Abstract: A method includes receiving an original string of bits where each of the bits represents one of two possible logic levels. The string of bits also carries information. A new string is formed, based on the original string, which contains all of the information of the original string by using fewer bits of one of the logic levels.

Abstract: A network controller for controlling a message over a communication network constantly monitors actually transmitted message from or to a particular user. If the controller determines a particular phrase is transmitted more than a predetermined number of times, the controller updates support data and transmits the data in a compressed format with support data. The network controller may send an updated support data in response to a request from a user equipment. In this way, the present invention reduces the amount of traffic over the network.

Abstract: A modem interface transfers data between a high data rate interface and a wireless interface. The modem interface has a plurality of parallel data highways. Each data highway has frames with time slots for transferring data. The plurality of highways outputs data to the high data rate interface and the wireless interface in selected time slots. At least one of the data highways has an input configured to receive data from the high data rate interface in selected time slots. At least one of the data highways has an input configured to receive data from the wireless interface in selected time slots. A processor controls the transfer of data between the plurality of highways.

Abstract: Real-time communication of multimedia data over heterogeneous networks that may include constant delay networks, variable delay networks that have a common reckoning of time, and variable delay networks that do not have a common reckoning of time. If there are any variable delay networks in which there is no common reckoning of time in the heterogeneous networks, a common reckoning of time is established in each of those networks. Then, a constant delay network is emulated for each variable delay network using the specific common time reckoning present in each variable delay network.

Abstract: An Ethernet mapping enables high speed Ethernet data streams having a data rate of 10 Gb/s to be transported across a synchronous packet switched network fabric having a standard SONET OC-192 line rate of 9.953280 Gbaud. The 10 Gb/s Ethernet data stream is compressed by removing interframe gaps between successive MAC frames to produce a compressed data stream, which is then mapped to a synchronous container. The synchronous container is then launched across the synchronous packet switched network fabric at a standard SONET OC-192 line rate of 9.953280 Gbaud. The synchronous container is preferably provided as a stripped STS-192c frame having only A1 and A2 octets of the Transport Overhead (TOH).

Abstract: Packet-based central conference bridges, packet-based network interfaces and packet-based terminals are used for voice communications over a packet-based network. Modifications to these apparatuses can reduce the latency and the signal processing requirements while increasing the signal quality within a voice conference as well as point-to-point communications. For instance, by selecting the talkers prior to the decompression of the voice signals, decreases in the latency and increases in signal quality within the voice conference can result due to a possible removal of the decompression and subsequent compression operations in a conference bridge unnecessary in some circumstances. Further, the removal of the jitter buffers within the conference bridges and the moving of the mixing operation to the individual terminals and/or network interfaces are modifications that can cause lower latency and transcoding within the voice conference.

Abstract: A data optimization engine disposed inline with a first communication channel and a second communication channel. The data optimization engine comprises a transmit interface circuit configured to receive a first data stream from the first communication channel and to obtain a first data file from the first data stream. The data optimization engine further includes an optimization processor coupled to the transmit interface circuit for receiving a second data file from the transmit interface circuit. The second data file represents the first data file after the first data file has been processed by the transmit interface circuit into a format suitable for optimization by the optimization processor. The optimization processor performs one of a compression and an encryption on the second data file, thereby obtaining all optimized data file. In one embodiment, the first data file is a Fiber Channel data frame.

Abstract: A method and apparatus are provided for exchanging information between at least some slots of a first T-carrier and some other non-coincidental slots of a second T-carrier. The method includes the steps of exchanging information between successive slots of the first T-carrier and respective predetermined memory locations within a memory device and exchanging information between successive slots of the second T-carrier and at least some of the predetermined locations in memory of the first T-carrier based upon a channel-exchange list relating at least some channels of the first T-carrier to at least some other channels of the second T-carrier.

Abstract: A proxy-server system (15) connected preferably to a computer-telephone system (10) intercepts, processes, and analyzes as traffic-analysis results (68A-C) all forms of real- and non-real-time electronic communication passing over the network in the form of raw traffic data (61). The proxy-server system normalizes each communication into the measure of time needed by recipient(s) of the communication to understand the information contained therein. Once normalized, the data may be aggregated into summary reports (69A-C). As part of the analysis, the aggregated communication records are compared with user-defined rules to provide alerts if the individual or aggregated durations exceed boundaries set by the rules. In one embodiment, the summary reports may be integrated with general-ledger data (94) and other raw business data (74) via a relational database (72) to derive more accurate records of activity-based-costing information (76).

Abstract: A method and apparatus for maintaining a plurality of sequence numbers in a plurality of header compressed packets associated with an information context is described. Header compressed packets are transferred on a communication link according to a multi-layer protocol having at least a higher layer and a lower layer. If it is determined that one or more header compressed packets has been dropped, a flag is set on the header compressed packets. A sequence count at the higher layer and lower layer may be reduced by a number corresponding to the number of header compressed packets dropped. The sequence number of each of the header compressed packets is checked at the lower layer against the sequence count and renumbered to match the sequence count. If the sequence number and the sequence count match the flag is cleared. The higher layer is a PPP layer and the lower layer is a Physical Layer.

Abstract: A UMTS (Universal Mobile Telecommunications System) core network supports a compression framework that provides for header compression of General Packet Radio Service Tunneling Protocol (GTP)-Encapsulated Packets. In particular; the GTP/UDP(User Datagram Protocol)/IP(Internet Protocol) header is compressed. In addition, the UMTS core network also supports RTP(Real Time Protocol)/UDP/IP header compression independent of the GTP/UDP/IP header compression.

Abstract: Data packets transmitted over a cable network are suppressed by hardware at the transmitting end and expanded on the receiving end. This conserves bandwidth as well as reduces the processing resource requirements in both the CM and the CMTS. An extended header element is added to a data packet that is to be transmitted over the cable network. The extended header element contains an index that is used along with the SID to access a rule. The rule is used to determine which bytes are to be suppressed at the transmitter and expanded at the receiver.

Abstract: A transcoder rate controller is provided which carries out tandem free operation (TFO) transcoding on inter-switching controller signals in a data transmission network.

Abstract: A method of transmitting signals along a communication path, by a first compression/decompression unit, which method comprises receiving first signals from a first direction along the path, determining based on the received first signals whether a second decompression unit is located along the path in the first direction, receiving second signals from a second direction. In the case it is determined that a second decompression unit is located along the path in the first direction, selecting a compressed or decompressed format for transmission of the second signals responsive to at least one additional parameter, such that the determination that a second decompression unit is located along the path in the first direction does not always bring about selecting the compressed format. Then, the second signals are transmitted in the selected format.

Abstract: A method for compressing data is provided that includes accumulating a plurality of bits associated with a communications flow and determining whether one or more of the bits correspond to a silence signal associated with a time division multiplexed (TDM) circuit that facilitates propagation of the flow. A predefined silence pattern may be communicated, in place of one or more of the bits, to a next destination when it is determined that one or more of the bits correspond to the silence signal.

Abstract: In a wireless packet switching telecommunications network, voice services are provided by having a compressor/decompressor in each mobile station to provide each voice packet with a compressed header. Voice data and signalling data are sent separately and in different data formats to the air interface. The compressed header may be an M bit and a cyclically reset timeclick_number, which is decompressed by use of a wallclock which counts reset cycles to reinstate the voice packet headers. Alternatively, RTP agents are provided at the compression and decompression points, and voice packets are sent without headers over a “high quality” network such as a frame relay or ATM network. Compression state of a voice packet header can be established by sending call set-up information over an out-of-band channel between compression points in a mobile station and in die network.

Abstract: A method for improving the sound quality of voice signals transmitted during a call between two digital wireless telephones by diverting the call from a circuit switched network to a data network is disclosed. In accordance with the present invention, a call setup procedure using the circuit switched network is modified so that the calling party may determine the following information: 1) whether the called party's digital wireless telephone is able to support one of the voice compression algorithms supported by the calling party's digital wireless telephone and 2) whether both wireless telephones have access to the same data network.

Abstract: Digital content is extracted from single and multiple-program transport streams for subsequent non-real-time (background) transmission in a bandwidth-efficient manner. At an encoder, several null and/or nonessential (N/N) packets (e.g., packets of unused programs) in a transport stream are replaced by an annotation packet (AP) to obtain non-real-time packets. The non-real time packets are inserted into null packet locations (NU) of a real time transport stream. At a decoder, the annotation packets are used to regenerate a specified number of null packets to maintain the original separation of packets of the non-real time data (for timing purposes), allowing existing hardware and/or software to decode the selected program.

Abstract: The invention relates to a method and apparatus for controlling the transition of a bypass capable codec between operative modes, based on a certain characteristic of the audio data signal processed by the codec. The apparatus relies on a control signal to determine when the codec will switch from one mode to another. This control signal reflects a characteristic of the audio data signal received at the apparatus, such as the type of speech activity or the format of the audio data signal. When in the active (non-bypass) mode, the apparatus relies on an additional control signal to switch to the inactive (bypass) mode. This additional control signal is received from a control unit at a remote codec that indicates that the remote codec is also bypass capable, hence the decoder at the first codec and the encoder at the remote codec can switch to the inactive mode to pass between them the compressed data frames.

Abstract: Update request signal generator 107 generates an update request signal when an NG signal is output from CRC section 104 a predetermined number of times, continuously. More specifically, update request signal generator 107 comprises a counter that counts the number of times errors are detected, increments the counter by 1 every time an NG signal is output from CRC section 104, and resets the counter when an OK signal is output from CRC section 104 or every time the value on the counter reaches a predetermined number of times. Then, update request signal generator 107 generates an update request signal when the value on the counter reaches the predetermined number of times.

Abstract: A bi-directional packet data transmission system for a packet data transmission between a terminal and a radio access network includes an uplink resource and a downlink resource which are independently set. Memory resources can be effectively managed even in a packet data transmission service with the asymmetrical structure such that the packet amount for the downlink is much greater than the packet amount for the uplink, or the packet amount for the uplink is much greater than the packet amount for the downlink.

Abstract: The present invention relates to a method and system for controlling header compression in a packet data network, e.g. an IP based cellular network. Load information of the packet data network is obtained, and header compression is triggered in response to the result of an evaluation of the load information. In particular, the header compression can be applied only for one direction, provided that the load information indicates asymmetry. Thereby, the headers can be compressed only on demand and only in the direction where the traffic volume is higher and where, in effect, the transmission capacity is the bottleneck. As a result, header compression takes less processing power and thus allows more flows to be compressed.

Abstract: The invention proposes a method for conveying data packets in a network, comprising the steps of examining (S1, S2) whether a received data packet is encrypted, compressing (S3) the data packet in case it is examined that the data packet is not encrypted, and refraining form compressing (S4) in case it is examined that the data packet is encrypted. The invention also proposes a corresponding network element.

Abstract: A method for transferring a data packet from a compressor to a decompressor said data packet including a header with header data fields. A number of the header data fields that remain constant during the data transfer are stored in the decompressor. In a compressed data packet, a header data field that varies is replaced by a compressed value identifying a data packet in a compression sequence. In the decompressor context data comprising information for relating the received compressed value to a corresponding compression sequence is maintained and the information is updated according to the received compressed values. The compressed value and the information of the corresponding compression sequence are used for mapping the compressed value into a decompressed header data field. Thus compressed data is unambiguously mapped to a full packet data header field in the decompressor side throughout the session.

Abstract: In a network system, clock signals of a server device and a client device are synchronized with each other without having any influence of a change and a delay of the transmission time of data.

Abstract: A method and a decompressor node in Internet Protocol (IP) network for fast change of IP Headers Compression (HC) mechanism are provided. A Mobile Station (MS) connected to a serving system uses a first IP HC mechanism therewith. Upon initiating handoff of the MS from the serving system to a target system, a second IP HC mechanism is chosen to be used with the target system. The second IP HC mechanism is initialized using information received at the target system from the serving system. In a preferred embodiment, the received information is a decompression context and the initialization of the second IP HC mechanism consists of translating the decompression context from the first IP HC mechanism to the second IP HC mechanism.

Abstract: Systems and methods for providing accelerated transmission of broadcast data, such as financial data and news feeds, over a communication channel using data compression and decompression to provide secure transmission and transparent multiplication of communication bandwidth, as well as reduce the latency associated with data transmission of conventional systems.

Abstract: A timer based header compression/decompression technique and timer and reference based technique are provided. A source generates a header field, such as an RTP time stamp. A packet including the header field is sent to a compressor which calculates a compressed header field based on the header field of the packet from the source and a jitter quantity. The compressed header field is calculated by calculating a jitter effect the network before the compressor has on the transmission of packets and calculating a jitter effect the network between the compressor and a decompressor has on the transmission of packets. The packet including the compressed header field is transmitted to a decompressor which includes a local timer. The decompressor decompresses the compressed header field by calculating an approximation of the header field based on the elapsed time since the arrival of a previous packet and the field value in the previous packet.

Abstract: Techniques are provided for efficiently compressing and reconstructing the time stamp value of a real time communications packet whose time stamp value does not fall within a normally expected sequence of time stamp values. A first part of the time stamp value is selected by the header compressor and transmitted. A second part of the time stamp value is estimated by the header decompressor based on elapsed time between receipt of consecutive packets. The header decompressor combines the second part with the first part received from the header compressor to produce a reconstructed time stamp value.

Abstract: The invention proposes a method and an apparatus for the radio transmission of user data, arranged in data words of a defined length, in time-division multiplex frames having active time slots in which data are transmitted and inactive time slots in which no data are transmitted. The user data are stored temporarily in a memory at a first data rate, and are read out from the memory and converted into the time-division multiplex frames at a second, greater data rate. During the time period from the beginning of an active time slot to the beginning of the subsequent active time slot, n (n=a whole number ≧1) data words of the user data are thereby respectively stored. By synchronizing the storing and the reading out of the user data, these data can be transmitted with the smallest possible delay in an arbitrarily selected time slot of the time-division multiplex frame.

Abstract: A system and method for adaptive voice buffering is disclosed. In a voice transmitter, a ring buffer uses read and write pointers to implement changes in the delay of a communication link. The receiver also implements a ring buffer having read and write pointers to buffer incoming digitized voice information transmitted over the link. Delay changes may be implemented during silent periods to render delay variations in the received signal undetectable to a user of the receiver.

Abstract: An Internet switch box connects between a telephone set and a public switched telephone network (PSTN) line, the latter of which is used both for PSTN telephone conversations and for connection to an Internet service provider (ISP). The switch box contains hardware and embedded software for establishing a connection to an ISP and for Internet telephony. When two users, each having an Internet switch box connected to the telephone set, wish to have an Internet telephony conversation, one calls the other over the PSTN. When they agree to an Internet telephony conversation, they signal their Internet switch boxes, by pressing either buttons on the switch boxes or certain keys on the telephone keypads, to switch to Internet telephony. The switch boxes disconnect the PSTN call and connect to their ISPs. Once the switch boxes are on the Internet, they contact each other through a server which supplies Internet protocol (IP) addresses of switch boxes, and the users continue their conversation by Internet telephony.

Abstract: Monitoring conditions within a switching network interconnecting two host systems engaged in a media call and adjusting the media parameters used by the host systems to improve call conditions.

Abstract: A method employing multisignal encoded start pulses and end pulses for data transmission by use of radio waves, carries out encoding so that a very large number of kinds of patterns (not less than 1014 kinds are possible) can be effected with one frequency. In the binary system, encoding needs a large number of signal pulses and high frequency, and the frequency of the carrier wave becomes very high also. The multisignal method of this invention makes it possible to reduce the encoding bits and the carrier wave frequency by using level sensing and width comparison techniques. This method also removes noise owing to its level sensing and width comparison techniques which have not been used in conventional digital communication systems. Therefore, the reception of erroneous information, the occurrence of unclear audio-visual images, and noise trouble are prevented in the use of radio waves.